1. Field of the Invention
This invention relates to multi-channel communications systems employing a redundancy removal scheme using predictive encoding of speech in a digital, multi-channel communications system for the purpose of bit rate reduction with no appreciable degradation in voice quality, and more particularly, to a frame synchronization technique for use in such communications systems without increasing the transmitted bit rate.
2. Description of the Prior Art
In communications system using long and expensive transmission facilities, such as submarines cables and satellite communications systems, terminal facilities which insure optimum utilization of the transmission channels are very important. A well-known analog system, the Time Assignment Speech Interpolation (TASI) system achieves communications efficiency, i.e. bandwidth compression, by means of a transmission time savings. The TASI system takes advantage of the statistical fact that during a telephone conversation a one-way telecommunications channel is active only on the average of about 50% of the time. The TASI system monitors each voice circuit for voice activity and, in response to the detection of voice, connects a talker to an available channel. In this manner, a number of talkers greater than the number of available channels may be serviced by sharing the channels on a talkspurt interpolated basis.
The quality of speech transmitted by TASI is effected by three main sources of degradation. First, degradation occurs due to interpolation. If the number of talkers simultaneously talking in one direction exceeds the number of available channels a certain number of these talkers will be temporarily denied a channel. This condition is known as "freeze-out". The portion of speech not being transmitted by a talker who is temporarily "frozen-out" results in speech quality degradation. Second, degradation occurs due to the operation time of speech detectors, there being one speech detector for each voice circuit. Prior to connecting a voice circuit to an available channel the voice detector must detect speech activity in the voice circuit. During the time required for the voice detector to actually detect voice, the talker's speech signals are lost causing further degradation of voice quality. Third, degradation is due to speech signals being lost during the time needed for switching and signaling functions to establish the proper connection between talker and listener once speech activity is detected by the voice detector.
There are many characteristics of the Speech Predictive Encoding Communication (SPEC) System as described in U.S. Pat. No. 3,927,268 which represent improvements over prior art TASI systems. These improvements, while mentioned here, will become more readily apparent from the detailed discussion of which follows. First the SPEC system achieves bandwidth reduction by accommodating the traffic of n telephone circuits in the capacity of n/2 telephone channels with no noticeable degradation in received voice quality. Secondly, the SPEC system, being an all digital system, makes decisions on each voice circuit at the basic sampling rate. For this reason, the transmission of data within the voice band, which is a difficult task for TASI-like systems, is easily accommodated. Third, the SPEC system employs a predictive encoding scheme which significantly reduces, by about 15%, the average activity factor (defined as the number of voice samples transmitted/the total number of voice samples) over prior art systems without any appreciable loss in voice quality. Fourth, whereas the effect of "freeze-out" in TASI-like systems manifests itself as a "chopping" or "clipping" of the voice signal which can result in the loss of an entire syllable, the effect of "overload" (i.e. freeze-out) in the SPEC system results only in an amplitude error (as opposed to a "clip") in the recieved voice signal. In an "overload" condition the SPEC system does not really "freeze-out" samples from the voice circuits "frozen-out" since those circuits will have corresponding voice samples stored at the receiver whereby the receiver can reconstruct replicas of the "frozen-out" samples. Also, by means of a recirculation of the servicing sequence of n voice circuits the subjective effect of "overload" is substantially reduced. Fifth, the SPEC system utilizes a parity check scheme for protecting the transmitted voice samples thereby reducing the effect of errors (resulting in small amplitude error) caused by channel noise. Sixth, the SPEC system is built in a modular configuration (i.e., 64 circuits serviced by 32 circuits) to permit easy expansion to large capacities. Seventh, the flexibility of the SPEC system allows transmission in either time division multiplex-frequency division multiple access (TDM-FDMA) or time division multiplex-time division multiple access (TDM-TDMA) systems. Eighth, the SPEC system can be used in a point to multipoint fashion in satellite communications. Any station can transmit voice information to several other stations while each of the other stations would use a reciever which only uses the specific voice circuits addressed to it. In this manner, larger amounts of telephone traffic destined for multiple stations can be interpolated at the transmitter of a single station. Finally, implementation of the SPEC system results in a lower cost per circuit as well as higher quality service than prior art systems such as TASI.
In the SPEC system, bit rate compression in a digital, multichannel, voice communications system is accomplished while maintaining normal voice transmission quality. The system is designed to transmit all information from n telephone circuits over the transmission capacity conventionally allocated for digital transmission of all voice information from n/2 circuits. All n voice circuits are sampled at a rate, known as the frame rate, of one voice circuit every 125.mu. secs. Each voice sample in a frame period is compared at the transmitter with the corresponding voice sample of a previous frame stored in a predictive frame memory (PFM). If the comparison indicates that the present sample is predictable from the corresponding previous sample, a logic "O" is generated indicating that the present sample need not be transmitted. If the comparison indicates that the present sample is unpredictable from the corresponding previous sample, then a logic "l" is generated indicating that the unpredictable sample should be transmitted.
Transmission of the unpredictable samples is accomplished in the following manner. A frame of information equivalent in bit rate to that required for conventional digital transmission of all voice information from n/2 voice circuits comprises the essential information and is formed at the transmitter. Assuming n = 64 the transmission frame comprises 24, 8 bit time slots T.sub.1 thru T.sub.24 designated for transmission of unpredictable samples and eight, 8 bit time slots T.sub.25 thru T .sub.32 occupied by a 64 bit sample assignment word (SAW). The SAW informs the receiver as to which of the 64 voice circuits the unpredictable samples T.sub.1 - T.sub.24 belong.
As the comparisons are made at the transmitter, the first comparison indicating an unpredictable sample results in that sample being placed in time slot T.sub.1. If that sample is from voice circuit 3, for example, then the SAW will have "O" in its first and second bit slots and a "1" in the third bit slot. If the next voice circuit indicative of unpredictability is, for example, voice circuit 6, then that unpredictable sample will be placed in time slot T.sub.2, and the SAW will have "O" bits in bit slots 4 and 5 and a "1" in bit slot 6. This operation continues until 64 comparisons have been made and the unpredictable samples placed in the available time slots T.sub.1 - T.sub.24.
The receiver already has stored therein 64 voice samples which were transmitted during previous frames as unpredictable samples. When the reciever receives the presently transmitted information including the sample assignment word, it then updates the corresponding 64 voice samples stored therein by substituting the unpredictable voice samples for the stored voice samples in accordance with the channel routing information provided by the SAW. The receiver is then in a position to properly reconstruct the present frame of all 64 voice samples.
The SPEC system is designed around the statistics of speech such that on the average in a system of 64 voice circuits of information, only 24 voice circuits will be non-redundant. However, there will be times when there is non-redundancy, i.e., unpredictability, in more than 24 voice circuits thereby resulting in an "overload" condition for those circuits which number above the 24 time slots available for transmission on that particular frame. The system alleviates "overload" in two ways. First, if an unpredictable sample is not transmitted because time slots T.sub.1 thru T.sub.24 are filled, the receiver utilizes the corresponding previous sample stored at the receiver for reconstruction of the unpredictable sample which could not be transmitted. Though the corresponding previous sample is being reconstructed as the unpredictable sample, the fact is the corresponding previous sample stored at the receiver should be close in value to the unpredictable sample which could not be transmitted. Secondly, the subjective effects of "overload" are alleviated by effectively recirculating the servicing sequence. For example, during frame 1 the voice circuits are serviced at the transmitter in sequence from 1 to 64. During the next frame, the voice circuits are effectively serviced in sequence starting with voice circuit 2; voice circuit 1 being the 64th circuit to be serviced; and so on. This recirculation of the servicing sequence continues so that in a period of 64 frames each circuit has had the opportunity to be serviced at each priority level (i.e. first to 64th). In this manner, if the system is operating under "overload" conditions the higher numbered circuits are not always serviced last since effectively those circuits become the lower numbered circuits on successive frames.
In order for the SPEC decoder to correctly reconstruct the PCM voice signals, the position of the first bit of each frame must be known. This is the classic problem of frame synchronization which is normally solved by adding synchronization information to each frame, which upon recognition at the receiver, automatically synchronizes the frame. This approach necessarily increases the transmission bit rate without increasing the information bit rate. In many cases, the synchronization scheme is required to achieve short acquisition time and low miss and false detection probabilities. As these requirements are made more stringent, the amount of synchronization information transmitted per frame must increase.